U.S. Pat. Nos. 5,583,962, 5,632,005 and 5,633,981 describe two reduced-bit-rate perceptual-coding systems for audio signals, designated therein as "Type I" and "Type II." Each of said U.S. Pat. Nos. 5,583,962, 5,632,005, and 5,633,981 patents is hereby incorporated by reference in its entirety. According to a principle underlying both systems, an encoder generates frequency subband signals in response to input audio signal streams, the subbands corresponding generally to the human ear's critical bands.
In the encoder of the Type I system described in said patents, each audio stream is encoded independently when there is a sufficient number of bits available. When there is a shortage of bits, the signal components in some or all of the subbands are combined into a composite signal and a plurality of scale factors, one scale factor for each input audio stream, each scale factor based on some measure of the subband signal components in each of the audio streams. The Type I decoder reconstructs a representation of the original signal streams from the composite signal and scale factors. The Type I system thus provides a bit savings or coding gain over a dedicated discrete system in which each audio stream is encoded independently. The Type I system is employed in AC-3 coding, which forms the basis of a the Dolby Digital perceptual coding system, in which 5.1 audio channels (left, center, right, left surround, right surround and a limited-bandwidth subwoofer channel) are encoded into a reduced bit-rate data stream.
In the encoder of the Type II system described in said patents, each audio stream is encoded independently when there is a sufficient number of bits available. When there is a shortage of bits, the signal components in some or all of the subbands are combined into a composite signal and one or more directional vectors, the directional vectors indicating the one or more principal directions of a soundfield represented by the audio streams. The Type II decoder reconstructs a representation of the soundfield represented by the original signal streams from the composite signal and the one or more directional vectors. The Type II system thus provides a bit savings or coding gain over a dedicated discrete system in which each audio stream is encoded independently and over the Type I system in which the composite signal is associated with scale factors for each audio stream.
The Type I and Type II systems described in said patents are adaptive in several ways. One aspect of their adaptivity is that one or more of the frequency subbands may operate some of the time in a "discrete" mode such that all of the subband components of the audio streams in the frequency subband are each independently encoded and decoded, while a shortage of bits, for example, causes the subband components of the audio streams in a particular frequency subband to be encoded according to the Type I approach or the Type II approach.
It is also known to change back and forth adaptively from a Type I to a Type II mode of operation within one or more frequency subbands. Such arrangements are the subject of the U.S. patent application of Mark Franklin Davis, Ser. No. 08/895,496, filed Jul. 16, 1997, entitled "Method and Apparatus for Encoding and Decoding Multiple Audio Channels at Low Bit Rates." Because the Type II approach requires fewer bits than does the Type I approach, a short term shortage of bits may be overcome by employing Type II encoding and decoding.